Even floating point numbers can clip and degrade.
If I'm mixing, I prefer 64bit floats if available, but I use mostly analogue gear now .. so the mixing happens outside of the digital domain.
I do have some software I wrote to automagically remix music.. the jazz-o-tron 1000. Internally it uses 80bit IEEE samples (YES, 80.. its not a typo).. but I only use ...
is ABSOLUTELY necessary to record at 48 kHz that's not for film where the final audio will be bounced to 44.1 format anyway?
We know that at 44.1kHz we can accurately record and playback the frequencies that live in the human hearing threshold, so oversampling might seem an overkill. Most of the time this is the case, but some scenarios can benefit from ...
It kinda matters what order you put them in but everyone has a different order that they like and then sometimes you adjust for issues. Most Channel strips let you change the order of at least some of the modules. I tend to default to EQ/DYN/De-es/multiband but everyone is different. Sometimes you might need 2 of certain processes, 1 to fix an issue, ...
As a possible side bar to the excellent answer provided by JCPedroza above,
If human ears are the only listeners (i.e., signals between 20 and 20,000 Hz) then it is never* absolutely necessary to record at 48kHz.
[H]e used an SM57 whose frequency range is 40 Hz to 15 kHz so it
wouldn't really make a difference because in the oversampling
Route the output of each track to another empty track. Record arm the destination tracks and then hit record. When the record pass is done all your effects and automation will be "printed" to the new tracks.
Avid provides replacement disks from their online store. You can access it here. You could try contacting them to see if there is a download option, but it is possible they may only provide physical media for boxed copies.
Izotope RX is probably what you'd want. You can start with basic if your on a budget and then upgrade to advanced later.
It has plugin components as well as a standalone version.
It and Cedar are pretty much the standards at this point.
Multiband compression/expansion or something like the Waves WNS or W43 can help for a constant noise.
I would apply them in order of de-esser, EQ and compressor. The first two could be done in either order, but the compressor should generally be last.
You could EQ with or without the De-esser applied, but the De-esser will offer you less control over the sound than a good EQ.
The compression should be last because it deals with overall signal power, which ...
It will only delete unused regions, meaning regions that are not used in any tracks in the edit window. So if you delete them, everything will play back as before, and you can continue editing.
The only reason you wouldn't want to delete these unused regions is if you may undo the editing that created them.
Some delay – proper word: latency – is inevitable whenever you monitor something through software: this always requires some digital data to be passed into the software, be processed, and passed out again – on general-purpose hardware that can only be done efficiently when you pass whole chunks of samples, usually something between 64 and 4096. ...
Yes, Pro Tools has something extremely similar.
Unlike Pro Tools, Logic does not have playlists. The easiest way to look at a 'playlist' is simply as a 'take'.
In Pro Tools, you can either record over the previously recorded region, or each recording can have its own playlist. Even if you didn't use playlists initially, it is easy to tell Pro Tools to ...
Personally I prefer Boar for version control of binary files. That is the exact use case of that system - mostly used for photos, etc.
On the other hand, I can also recommend you the git Large File System. That means you can stay with git syntax/toolchain without a need of setting up boar.
Took the above advice. This has been working great for me with git. Logic Pro X 10+.
Project\ File\ Backups
Haven't tried it, but splice looks interesting as well.
I'm not sure I follow your question 100%. Is this to picture? If so, and maybe even if not, I would bounce down my music to individual stems. That way, the music's tempo changes, quantization, reverbs, eq's, and whatever else is already taken care of. Then you can pull those stems into a session that has a constant "tempo" (framerate, if to picture). As ...
If you hit cmd and 4 on the neumeric keypad (ctrl 4 on PC) or go to Window>Automation the automation window will pop up. From there just hit the 'Suspend' button at the top and all automation in the session will be suspended.
I think all ProTools plugins have a master bypass that you can automate to bypass the plugin. I don't know that it really saves power though.
ProTools 11 has dynamic plugin allocation so the plugs that aren't processing aren't taking up power.
Other options might be thinning the automation more or increasing your playback buffer size.
"Option -" (minus sign) Shows all volume automation. Cntrl+Command Left and Right Arrow keys goes through all the automation paramters on a selected track. And Cntrl Option Cmnd left and right on arrow keys goes through all the automation paramters on ALL tracks.
John Purcell is the man when it comes to dialogue editing. He has some great videos which are free to watch on youtube, and fortunately for you I just happen to have them all in a playlist...
I'll bring them all in and sync them first, then do a broad chop of all of the unusable stuff. This is so that I can cover those tracks with crossfades, which I can't do in RX. Once I have my basic edits completed I'll do an RX pass, then reimport to protools for mixdowns.
My advice would be to always import everything into PT, make your choice which 'blend' of mics suit each other over the edits.
Then audiosuite/RX anything needed, but keep the original in place, muted, or on an inactive track.
My personal experience is to go as easy on the denoiser as possible, and use volume and eq to fix up things.
Happy mixing :-)
You should drop the buffer size from your sound card settings until the point your CPU can handle. Aim for less than 10ms of latency. For example a guitar player standing 3m away from the amplifier experiences a 10ms delay. 3ms should be quite unnoticable.
If you are not applying any insert effects you can use direct monitoring for 0 latency. But even with ...
Pro Tools 11 does not use .dpm plugins.
It only uses 64bit AAX plugins. If you log into native instruments service center, it will show you all available updates for your registered software. This includes updates to Kontakt 5 that provide the necessary 64bit AAX plugin format for use with Pro Tools 11.
The NI Service Center Updater is not smart enough to ...
I'm not a regular pro-tools user, but a universal way I'd approach this in any DAW is to:
Open a new project with the sample in question.
Create an automation for the volume of the sample.
Add control points to the automation as required.
Render the stem to another sample. This could be achieved by simply rendering the project.
Optionally you can skip the ...
ProTools does have a built in surround panner, it has for a long time.
Have you read the manual at all? It sounds like you already have a bias against ProTools without having actually used it all based on your 2nd question.
You just have to have the 5.1 mixer installed and surround panning will work.
Did you click on the small fader next to the volume ...
I currently go to the undergrad program at SCAD.
Besides the faculty and classes, it's a great place to network with other art majors. Because we're in high demand, you often times get the pick of the litter working on animations, films, motion media, ect.
For a number of alumni, they've gotten a lot of jobs from their SCAD connections
The Master's ...